#Designing a realtime audio/video conferencing service
Features
Functionality to consider when considering tool/platform.
- topology
- mesh (P2P)
all processing at end-points, participants limited by bandwidth
- routing (SFU)
light server processing, participants limited by bandwidth
- mixing (MCU)
heavy server processing
- stream efficiency - recipient
- Support suspending select streams at receiving end
i.e. pause reception of stream (not just omit further processing it)
- stream efficiency - forwarding
- Support simulcast
i.e. encode multiple streams that an SFU can "hop" between
- Support spatial/temporal/quality SVC
i.e. encode a stream that an SFU can efficiently "slice" without recoding
- stream efficiency - source
- stream efficiency - conference admin or shared room setting
- Skip video streams beyond a threshold of participants
- Skip video streams tied to quiet audio streams
- Skip streams of explicitly tagged non-speaker participants
- Mix audio streams (not stream each individually)
- security
- Support PERC
- Support ZRTP
i.e. end-to-end encryption (not only client-to-server)
- meeting management
- Personalized Meeting rooms
- Scheduled/Meet-me Meetings
- Instant/Direct Meetings
- Presence Support
- Recording
- Text chat
- Screen sharing
- feedback on own audio level
- feedback on encoding and streaming qualities
- conference management
- Conference Recording
- force-mute participants
- "Raise a hand" for muted participants
- meeting room
- Dual stream for dual screen
- Dial in from telephone
- Dial in from SIP audio-only
- Dial in from SIP with video
- Dial in from SIP with SIMPLE text chat
See also
SFU
Janus Gateway
WebRTC SFU/bridge/broker
written in C
Mediasoup
WebRTC SFU
written in C
Medooze SFU
WebRTC SFU
written in Node.js
Spreed WebRTC
WebRTC SFU
written in NodeJS and Go
Jitsi Videobridge
XMPP SFU
written in Java
seems to require "plan B" SDP: https://github.com/jitsi/jitsi-meet/issues/4758
MCU
Kurento
WebRTC MCU
written in C++
Licode
WebRTC MCU
written in C++
Medooze WebRTC Media Server
WebRTC/SIP MCU
written in C++
Red5 server
WebRTC SFU
written in Java
Ant Media Server
WebRTC SFU
written in Java
(fork of Red5 server)
Bridge/management
SylkServer
SIP/XMPP Application Server
using Janus
written in Python
Jigasi
WebRTC bridge to Jitsi Videobridge
written in Java
Matrix
using Jigasi and Jitsi Videobridge
BigBlueButton
using Kurento
written in Java
OpenMeetings
written in Java
Wire
proprietary-protocol Free Software stack
written in Haskell, Rust, C
Web frontend
multiparty-meeting
using Mediasoup
(and optionally drachtio and Kurento)
written in JavaScript
hosted at https://letsmeet.no/
Jangouts
using Janus
written in CoffeeScript
tawk.space
using Janus
written in CoffeeScript
hosted at https://tawk.space/
SIP2SIP
using SylkServer and Janus
hosted at https://sip2sip.info/
and https://webrtc.sipthor.net/
Roll Call
audio-only
hosted at https://roll.call
Spreed.ME
using Spreed WebRTC
Nextcloud Talk
using Spreed WebRTC
Jitsi Meet
using Jigasi and Jitsi Videobridge
hosted at https://meet.jit.si/
mConf
using Kurento
written in Java and Ruby
Veeting
cloud SFU service
using Janus
hosted at https://rooms.veeting.com/
(free trial at https://rooms.veeting.com/home-office)
Talky
cloud SFU service
hosted at https://talky.io/
Me
cloud SFU service
GoToMeeting
cloud SFU service
Zoom Meetings
cloud SFU service
supporting "up to 50 participants at once"
(but client bandwidth and resource demands and stability of such session is unknown)
Hangouts Meet
cloud SFU service
Webex Meetings
cloud SFU service
Wowza Streaming Cloud
cloud streaming service
Skype
cloud SFU service
suporting "up to 25 participants at once"
(but client bandwidth and resource demands and stability of such session is unknown)
MoxieMeet
cloud SFU service
requiring Google account
supporting "up to 32 users all on video together"
(but client bandwidth and resource demands and stability of such session is unknown)
TeamViewer
cloud SFU service
